Hello,
I'm currently trying to setup a SIP environment for VoIP calling for my final school project, so I'm just working with VoIP/SIP for 2 weeks.
I'm using SER as a SIP proxy server, but the carrier/gateway I am using for calling to/from PSTN is requiring me to register at their server and authorize outgoing calls, which is something SER won't do. So I got the idea to use asterisk between the PSTN carrier and SER for the authorization, since Asterisk can register and auth itself.
SIP --------- SIP --------- SIP --------- PSTN -----| |-----------| |-----------| |------ --------- --------- --------- SER Asterisk Carrier <-- auth stuff --> <-- sip relay -->
Has anyone here ever tried a similar setup or is this an impossibility ?
Kind regards,
E. Versaevel
If you are using asterisk just for the purpose of registering with the gateway then you could also consider using sipsak for that (that should be simpler).
sipsak can send the registration and you can configured it to put the IP address of your proxy server into the Contact.
Jan.
On 12-11 15:03, E. Versaevel wrote:
Hello,
I'm currently trying to setup a SIP environment for VoIP calling for my final school project, so I'm just working with VoIP/SIP for 2 weeks.
I'm using SER as a SIP proxy server, but the carrier/gateway I am using for calling to/from PSTN is requiring me to register at their server and authorize outgoing calls, which is something SER won't do. So I got the idea to use asterisk between the PSTN carrier and SER for the authorization, since Asterisk can register and auth itself.
SIP --------- SIP --------- SIP --------- PSTN -----| |-----------| |-----------| |------ --------- --------- --------- SER Asterisk Carrier <-- auth stuff --> <-- sip relay -->
Has anyone here ever tried a similar setup or is this an impossibility ?
Kind regards,
E. Versaevel
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
I have to authenticate on each outgoing call, so I've setup SER+Asterisk, however I still have some problems. I already asked this on the asterisk mailing list, but I'm not getting any replies there.
Ser forwards outgoing calls to asterisk, which authenticates itself to the carrier proxy, so far so good. However, asterisk has to forward incoming calls from the carrier to SER for routing to the UA's, however, asterisk alters the from header in the sip messages, this way when someone adds the incoming call to the address book, the sip uri is wrong and the call won't get anywere.
Take a look at this sip msg:
INVITE sip:erik@localphone:5061 SIP/2.0 Max-Forwards: 10 Record-Route: sip:3400009521@ser.box;ftag=as3f718642;lr=on Via: SIP/2.0/UDP ser.box;branch=z9hG4bK93dc.71ad80b3.0 Via: SIP/2.0/UDP ser.box:5065;branch=z9hG4bK513b584d From: "3400009525" sip:asterisk@ser.box:5065;tag=as3f718642 To: sip:3400009521@sip.ser.box Contact: sip:asterisk@ser.box:5065 Call-ID: 533cb84a48058ebb71fbd7bf7557c0f0@ser.box CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 18 Nov 2004 12:46:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 220 P-hint: USRLOC
As you can see the from user is not correct, this should be 3400009525@iptel.org. If a user adds this entry to a phonebook, the contact info will be wrong. For outgoing calls this won't be a problem as I only send calls to the pstn thru asterisk, incoming is a bit troublesome.
-----Oorspronkelijk bericht----- Van: Jan Janak [mailto:jan@iptel.org] Verzonden: zondag 14 november 2004 13:14 Aan: E. Versaevel CC: serusers@lists.iptel.org Onderwerp: Re: [Serusers] SER Proxy auth
If you are using asterisk just for the purpose of registering with the gateway then you could also consider using sipsak for that (that should be simpler).
sipsak can send the registration and you can configured it to put the IP address of your proxy server into the Contact.
Jan.
On 12-11 15:03, E. Versaevel wrote:
Hello,
I'm currently trying to setup a SIP environment for VoIP calling for my final school project, so I'm just working with VoIP/SIP for 2 weeks.
I'm using SER as a SIP proxy server, but the carrier/gateway I am using
for
calling to/from PSTN is requiring me to register at their server and authorize outgoing calls, which is something SER won't do. So I got the
idea
to use asterisk between the PSTN carrier and SER for the authorization, since Asterisk can register and auth itself.
SIP --------- SIP --------- SIP --------- PSTN -----| |-----------| |-----------| |------ --------- --------- --------- SER Asterisk Carrier <-- auth stuff --> <-- sip relay -->
Has anyone here ever tried a similar setup or is this an impossibility ?
Kind regards,
E. Versaevel
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Hello.
The other day I asked about video speed in sessions. Now I have seen that although I can establish video sessions between UAs, I get this error:
"error mediaproxy/getSDPMessage(): SDP message has zero length. use_media_proxy();failed to get the SDP message"
I mean, we can see and hear each other ( with really really bad video ) but I have these errors.
I have seen with ethereal that the INVITE packets carry an SDP packet too, so I don't understand why mediaproxy can not extract SDP packet. Maybe this is why video quality is so bad ? Does anyone know why this is happening ?
Thanks in advance:
Kiko
OK, I'm a bit further now, my problem only occurs when using non numeric usernames, ie if I use 1234@host it remains 1234 but if I use erik@host it turns into asterisk@host
-----Oorspronkelijk bericht----- Van: serusers-bounces@iptel.org [mailto:serusers-bounces@lists.iptel.org] Namens E. Versaevel Verzonden: maandag 22 november 2004 9:58 Aan: 'Jan Janak' CC: serusers@lists.iptel.org Onderwerp: RE: [Serusers] SER Proxy auth
I have to authenticate on each outgoing call, so I've setup SER+Asterisk, however I still have some problems. I already asked this on the asterisk mailing list, but I'm not getting any replies there.
Ser forwards outgoing calls to asterisk, which authenticates itself to the carrier proxy, so far so good. However, asterisk has to forward incoming calls from the carrier to SER for routing to the UA's, however, asterisk alters the from header in the sip messages, this way when someone adds the incoming call to the address book, the sip uri is wrong and the call won't get anywere.
Take a look at this sip msg:
INVITE sip:erik@localphone:5061 SIP/2.0 Max-Forwards: 10 Record-Route: sip:3400009521@ser.box;ftag=as3f718642;lr=on Via: SIP/2.0/UDP ser.box;branch=z9hG4bK93dc.71ad80b3.0 Via: SIP/2.0/UDP ser.box:5065;branch=z9hG4bK513b584d From: "3400009525" sip:asterisk@ser.box:5065;tag=as3f718642 To: sip:3400009521@sip.ser.box Contact: sip:asterisk@ser.box:5065 Call-ID: 533cb84a48058ebb71fbd7bf7557c0f0@ser.box CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 18 Nov 2004 12:46:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 220 P-hint: USRLOC
As you can see the from user is not correct, this should be 3400009525@iptel.org. If a user adds this entry to a phonebook, the contact info will be wrong. For outgoing calls this won't be a problem as I only send calls to the pstn thru asterisk, incoming is a bit troublesome.
-----Oorspronkelijk bericht----- Van: Jan Janak [mailto:jan@iptel.org] Verzonden: zondag 14 november 2004 13:14 Aan: E. Versaevel CC: serusers@lists.iptel.org Onderwerp: Re: [Serusers] SER Proxy auth
If you are using asterisk just for the purpose of registering with the gateway then you could also consider using sipsak for that (that should be simpler).
sipsak can send the registration and you can configured it to put the IP address of your proxy server into the Contact.
Jan.
On 12-11 15:03, E. Versaevel wrote:
Hello,
I'm currently trying to setup a SIP environment for VoIP calling for my final school project, so I'm just working with VoIP/SIP for 2 weeks.
I'm using SER as a SIP proxy server, but the carrier/gateway I am using
for
calling to/from PSTN is requiring me to register at their server and authorize outgoing calls, which is something SER won't do. So I got the
idea
to use asterisk between the PSTN carrier and SER for the authorization, since Asterisk can register and auth itself.
SIP --------- SIP --------- SIP --------- PSTN -----| |-----------| |-----------| |------ --------- --------- --------- SER Asterisk Carrier <-- auth stuff --> <-- sip relay -->
Has anyone here ever tried a similar setup or is this an impossibility ?
Kind regards,
E. Versaevel
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
_______________________________________________ Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
I am not sure I understand your description and the problem. If SER gets the INVITE then registration with the PSTN gateway works (I think that was the original problem).
Jan.
On 22-11 09:58, E. Versaevel wrote:
I have to authenticate on each outgoing call, so I've setup SER+Asterisk, however I still have some problems. I already asked this on the asterisk mailing list, but I'm not getting any replies there.
Ser forwards outgoing calls to asterisk, which authenticates itself to the carrier proxy, so far so good. However, asterisk has to forward incoming calls from the carrier to SER for routing to the UA's, however, asterisk alters the from header in the sip messages, this way when someone adds the incoming call to the address book, the sip uri is wrong and the call won't get anywere.
Take a look at this sip msg:
INVITE sip:erik@localphone:5061 SIP/2.0 Max-Forwards: 10 Record-Route: sip:3400009521@ser.box;ftag=as3f718642;lr=on Via: SIP/2.0/UDP ser.box;branch=z9hG4bK93dc.71ad80b3.0 Via: SIP/2.0/UDP ser.box:5065;branch=z9hG4bK513b584d From: "3400009525" sip:asterisk@ser.box:5065;tag=as3f718642 To: sip:3400009521@sip.ser.box Contact: sip:asterisk@ser.box:5065 Call-ID: 533cb84a48058ebb71fbd7bf7557c0f0@ser.box CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 18 Nov 2004 12:46:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 220 P-hint: USRLOC
As you can see the from user is not correct, this should be 3400009525@iptel.org. If a user adds this entry to a phonebook, the contact info will be wrong. For outgoing calls this won't be a problem as I only send calls to the pstn thru asterisk, incoming is a bit troublesome.
-----Oorspronkelijk bericht----- Van: Jan Janak [mailto:jan@iptel.org] Verzonden: zondag 14 november 2004 13:14 Aan: E. Versaevel CC: serusers@lists.iptel.org Onderwerp: Re: [Serusers] SER Proxy auth
If you are using asterisk just for the purpose of registering with the gateway then you could also consider using sipsak for that (that should be simpler).
sipsak can send the registration and you can configured it to put the IP address of your proxy server into the Contact.
Jan.
On 12-11 15:03, E. Versaevel wrote:
Hello,
I'm currently trying to setup a SIP environment for VoIP calling for my final school project, so I'm just working with VoIP/SIP for 2 weeks.
I'm using SER as a SIP proxy server, but the carrier/gateway I am using
for
calling to/from PSTN is requiring me to register at their server and authorize outgoing calls, which is something SER won't do. So I got the
idea
to use asterisk between the PSTN carrier and SER for the authorization, since Asterisk can register and auth itself.
SIP --------- SIP --------- SIP --------- PSTN -----| |-----------| |-----------| |------ --------- --------- --------- SER Asterisk Carrier <-- auth stuff --> <-- sip relay -->
Has anyone here ever tried a similar setup or is this an impossibility ?
Kind regards,
E. Versaevel
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
The registration/authentication to the gateway works fine, the problem was/is that asterisk changes the From: username from erikje@iptel.org to asterisk@ser.box however, asterisk only does that with non numeric usernames, so 3400009525@iptel.org gets changed in 3400009525@ser.box, which is not a problem. So I just have to use numeric usernames :)
Kind regards,
E. Versaevel
-----Oorspronkelijk bericht----- Van: 'Jan Janak' [mailto:jan@iptel.org] Verzonden: donderdag 25 november 2004 20:48 Aan: E. Versaevel CC: serusers@lists.iptel.org Onderwerp: Re: [Serusers] SER Proxy auth
I am not sure I understand your description and the problem. If SER gets the INVITE then registration with the PSTN gateway works (I think that was the original problem).
Jan.
On 22-11 09:58, E. Versaevel wrote:
I have to authenticate on each outgoing call, so I've setup SER+Asterisk, however I still have some problems. I already asked this on the asterisk mailing list, but I'm not getting any replies there.
Ser forwards outgoing calls to asterisk, which authenticates itself to the carrier proxy, so far so good. However, asterisk has to forward incoming calls from the carrier to SER for routing to the UA's, however, asterisk alters the from header in the sip messages, this way when someone adds the incoming call to the address book, the sip uri is wrong and the call won't get anywere.
Take a look at this sip msg:
INVITE sip:erik@localphone:5061 SIP/2.0 Max-Forwards: 10 Record-Route: sip:3400009521@ser.box;ftag=as3f718642;lr=on Via: SIP/2.0/UDP ser.box;branch=z9hG4bK93dc.71ad80b3.0 Via: SIP/2.0/UDP ser.box:5065;branch=z9hG4bK513b584d From: "3400009525" sip:asterisk@ser.box:5065;tag=as3f718642 To: sip:3400009521@sip.ser.box Contact: sip:asterisk@ser.box:5065 Call-ID: 533cb84a48058ebb71fbd7bf7557c0f0@ser.box CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 18 Nov 2004 12:46:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 220 P-hint: USRLOC
As you can see the from user is not correct, this should be 3400009525@iptel.org. If a user adds this entry to a phonebook, the
contact
info will be wrong. For outgoing calls this won't be a problem as I only send calls to the
pstn
thru asterisk, incoming is a bit troublesome.
-----Oorspronkelijk bericht----- Van: Jan Janak [mailto:jan@iptel.org] Verzonden: zondag 14 november 2004 13:14 Aan: E. Versaevel CC: serusers@lists.iptel.org Onderwerp: Re: [Serusers] SER Proxy auth
If you are using asterisk just for the purpose of registering with the gateway then you could also consider using sipsak for that (that should be simpler).
sipsak can send the registration and you can configured it to put the IP address of your proxy server into the Contact.
Jan.
On 12-11 15:03, E. Versaevel wrote:
Hello,
I'm currently trying to setup a SIP environment for VoIP calling for my final school project, so I'm just working with VoIP/SIP for 2 weeks.
I'm using SER as a SIP proxy server, but the carrier/gateway I am using
for
calling to/from PSTN is requiring me to register at their server and authorize outgoing calls, which is something SER won't do. So I got the
idea
to use asterisk between the PSTN carrier and SER for the authorization, since Asterisk can register and auth itself.
SIP --------- SIP --------- SIP --------- PSTN -----| |-----------| |-----------| |------ --------- --------- --------- SER Asterisk Carrier <-- auth stuff --> <-- sip relay -->
Has anyone here ever tried a similar setup or is this an impossibility ?
Kind regards,
E. Versaevel
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-
'Jan Janak' -
E. Versaevel -
Jan Janak
-
Kiko Vives